kjellander 602f41e2ed Revert of Set defines for Chromium build. (patchset #3 id:40001 of https://codereview.webrtc.org/1847013002/ )
Reason for revert:
This breaks remoting_unittests on Windows in Chromium:
[5116:2536:0404/012329:5457156:ERROR:webrtcsession.cc(1388)] ConnectDataChannel called when data_channel_ is NULL.
[5116:2536:0404/012329:5457187:ERROR:opensslidentity.cc(154)] Generating certificate: error:0c000071:ASN.1 encoding routines:OPENSSL_internal:ERROR_GETTING_TIME

[5116:2536:0404/012329:5457218:ERROR:opensslidentity.cc(154)] Generating certificate: error:0c000071:ASN.1 encoding routines:OPENSSL_internal:ERROR_GETTING_TIME

[5116:2536:0404/012329:5457218:WARNING:dtlsidentitystore.cc(221)] Failed to generate DTLS identity.
[

Original issue's description:
> Set defines for Chromium build.
>
> Copy the defines from the target_defaults section of Chromium's
> src/third_party/libjingle.gyp into our webrtc/build/common.gypi
> in order to ensure the same defines are used for the Chromium build
> when removing the source listings in src/third_party/libjingle.gyp.
> With this CL landed, it should be possible to replace them with
> dependencies on:
> * webrtc/api/api.gyp:libjingle_peerconnections
> * webrtc/media/media.gyp:rtc_media
> * webrtc/pc/pc.gyp:rtc_pc
> * webrtc/pp2/p2p.gyp:rtc_p2p
> * webrtc/libjingle/xmpp/xmpp.gyp:rtc_xmpp
>
> Not ported (Windows specific):
> * Precompiled headers (build/win_precompile.gypi):
>   since it only seems to offer a compile speedup. Will be landed
>   for all of WebRTC in separate CL.
>
> BUG=webrtc:4256
> NOTRY=True
> R=perkj@webrtc.org, tommi@webrtc.org
>
> Committed: 9266cc0668

TBR=perkj@webrtc.org,tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4256

Review URL: https://codereview.webrtc.org/1861603002

Cr-Commit-Position: refs/heads/master@{#12229}
2016-04-05 06:39:51 +00:00
2014-06-17 08:54:03 +00:00
2015-09-11 09:04:09 +00:00

WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.

Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.

The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others. This page is maintained by the Google Chrome team.

Development

See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.

More info

Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
Readme BSD-3-Clause 446 MiB
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